In the case of the so-called “voice-over-IP telephony”, private branch exchanges or other technical devices such as for example private exchanges or public exchanges are coupled by means of special gateways via the Internet or an intranet in such a way that it is possible to make a call from a conventional (circuit-switched) telephone or a telecommunication terminal to another conventional telephone or an IP telephone (packet-switched terminal device) by using the “normal” telephone numbers or the call numbers.
FIG. 1 shows a simplified block diagram of a telecommunication system for the implementation of such a “voice-over-IP telephony”, in accordance with the prior art, it for example being possible that conventional telecommunication terminal devices TE1 and TE2 are integrated into a circuit-switched network, such as for example a conventional network with a public exchange (PSTN, Public Switched Telephone Network) or a private exchange.
To implement signaling adaptation, provision is made for a signaling adapter A in accordance with FIG. 1, which for example converts the call number TNe generated by pulse dialing or tone dialing (DTMF, Dual Tone Multiple Frequency) into an IP address IP-A. With such an IP address a connection setup or a signaling can be carried out in a packet-switched network N, such as for example the Internet or an intranet up to an IP terminal device IP-TE, which for example has a personal computer PC and has an input/output unit EA connected thereto. A connection between the two telecommunication terminal devices IP-TE and the signaling adapter A to the packet-switched network N is made for example by means of a DSL line (Digital Subscriber Line).
In the case of such adapters which make further connections over Internet protocols (for example, SIP, Session Initiated Protocol), as are implemented for example in the switching systems or private branch exchanges, a telephone number selected by a connected telecommunication terminal device TE1 or TE2 or a call number TNe must be completely known before the start of the Internet-side protocol setup, since the interpretation is not performed digit-by-digit by the exchange itself or by superordinate exchanges, as is the case in the conventional circuit-switched network. That means that the exchange itself does not have a criterion by which it recognizes a last digit of a selected digit sequence for a call number TNe of a connected telecommunication terminal device TE1 or TE2. This applies both to pulse dialing, by means of which the digits are produced by a plurality of pulses, and to a tone dialing (DTMF, Dual Tone Multiple Frequency), by means of which a predetermined frequency is produced and transferred for each digit.